Merge tag 'v3.12' into drm-intel-next

I want to merge in the new Broadwell support as a late hw enabling
pull request. But since the internal branch was based upon our
drm-intel-nightly integration branch I need to resolve all the
oustanding conflicts in drm/i915 with a backmerge to make the 60+
patches apply properly.

We'll propably have some fun because Linus will come up with a
slightly different merge solution.

Conflicts:
	drivers/gpu/drm/i915/i915_dma.c
	drivers/gpu/drm/i915/i915_drv.c
	drivers/gpu/drm/i915/intel_crt.c
	drivers/gpu/drm/i915/intel_ddi.c
	drivers/gpu/drm/i915/intel_display.c
	drivers/gpu/drm/i915/intel_dp.c
	drivers/gpu/drm/i915/intel_drv.h

All rather simple adjacent lines changed or partial backports from
-next to -fixes, with the exception of the thaw code in i915_dma.c.
That one needed a bit of shuffling to restore the intent.

Oh and the massive header file reordering in intel_drv.h is a bit
trouble. But not much.

v2: Also don't forget the fixup for the silent conflict that results
in compile fail ...

Signed-off-by: Daniel Vetter <daniel.vetter@ffwll.ch>

29 files changed, 157 insertions, 67 deletions

diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 17f45e8aa89..e1e9e0c999f 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -49,6 +49,8 @@ static struct snd_pcm *snd_pcm_get(struct snd_card *card, int device)
 	struct snd_pcm *pcm;
 
 	list_for_each_entry(pcm, &snd_pcm_devices, list) {
+		if (pcm->internal)
+			continue;
 		if (pcm->card == card && pcm->device == device)
 			return pcm;
 	}
@@ -60,6 +62,8 @@ static int snd_pcm_next(struct snd_card *card, int device)
 	struct snd_pcm *pcm;
 
 	list_for_each_entry(pcm, &snd_pcm_devices, list) {
+		if (pcm->internal)
+			continue;
 		if (pcm->card == card && pcm->device > device)
 			return pcm->device;
 		else if (pcm->card->number > card->number)
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 445ca481d8d..bf578ba2677 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -175,6 +175,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
 { 0x54524106, 0xffffffff, "TR28026",		NULL,		NULL },
 { 0x54524108, 0xffffffff, "TR28028",		patch_tritech_tr28028,	NULL }, // added by xin jin [07/09/99]
 { 0x54524123, 0xffffffff, "TR28602",		NULL,		NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)]
+{ 0x54584e03, 0xffffffff, "TLV320AIC27",	NULL,		NULL },
 { 0x54584e20, 0xffffffff, "TLC320AD9xC",	NULL,		NULL },
 { 0x56494161, 0xffffffff, "VIA1612A",		NULL,		NULL }, // modified ICE1232 with S/PDIF
 { 0x56494170, 0xffffffff, "VIA1617A",		patch_vt1617a,	NULL }, // modified VT1616 with S/PDIF
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 5b6c4e3c92c..748c6a94196 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -4864,8 +4864,8 @@ static void hda_power_work(struct work_struct *work)
 	spin_unlock(&codec->power_lock);
 
 	state = hda_call_codec_suspend(codec, true);
-	codec->pm_down_notified = 0;
-	if (!bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) {
+	if (!codec->pm_down_notified &&
+	    !bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) {
 		codec->pm_down_notified = 1;
 		hda_call_pm_notify(bus, false);
 	}
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index ac41e9cdc97..b7c89dff706 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -3531,7 +3531,7 @@ static int create_capture_mixers(struct hda_codec *codec)
 		if (!multi)
 			err = create_single_cap_vol_ctl(codec, n, vol, sw,
 							inv_dmic);
-		else if (!multi_cap_vol)
+		else if (!multi_cap_vol && !inv_dmic)
 			err = create_bind_cap_vol_ctl(codec, n, vol, sw);
 		else
 			err = create_multi_cap_vol_ctl(codec);
@@ -4475,9 +4475,11 @@ int snd_hda_gen_build_controls(struct hda_codec *codec)
 					    true, &spec->vmaster_mute.sw_kctl);
 		if (err < 0)
 			return err;
-		if (spec->vmaster_mute.hook)
+		if (spec->vmaster_mute.hook) {
 			snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute,
 						 spec->vmaster_mute_enum);
+			snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
+		}
 	}
 
 	free_kctls(spec); /* no longer needed */
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 0cbdd87dde6..2aa2f579b4d 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -968,6 +968,15 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec,
 	}
 }
 
+static void ad1884_fixup_thinkpad(struct hda_codec *codec,
+				  const struct hda_fixup *fix, int action)
+{
+	struct ad198x_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		spec->gen.keep_eapd_on = 1;
+}
+
 /* set magic COEFs for dmic */
 static const struct hda_verb ad1884_dmic_init_verbs[] = {
 	{0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
@@ -979,6 +988,7 @@ enum {
 	AD1884_FIXUP_AMP_OVERRIDE,
 	AD1884_FIXUP_HP_EAPD,
 	AD1884_FIXUP_DMIC_COEF,
+	AD1884_FIXUP_THINKPAD,
 	AD1884_FIXUP_HP_TOUCHSMART,
 };
 
@@ -997,6 +1007,12 @@ static const struct hda_fixup ad1884_fixups[] = {
 		.type = HDA_FIXUP_VERBS,
 		.v.verbs = ad1884_dmic_init_verbs,
 	},
+	[AD1884_FIXUP_THINKPAD] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = ad1884_fixup_thinkpad,
+		.chained = true,
+		.chain_id = AD1884_FIXUP_DMIC_COEF,
+	},
 	[AD1884_FIXUP_HP_TOUCHSMART] = {
 		.type = HDA_FIXUP_VERBS,
 		.v.verbs = ad1884_dmic_init_verbs,
@@ -1008,7 +1024,7 @@ static const struct hda_fixup ad1884_fixups[] = {
 static const struct snd_pci_quirk ad1884_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x2a82, "HP Touchsmart", AD1884_FIXUP_HP_TOUCHSMART),
 	SND_PCI_QUIRK_VENDOR(0x103c, "HP", AD1884_FIXUP_HP_EAPD),
-	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_DMIC_COEF),
+	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_THINKPAD),
 	{}
 };
 
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 4edd2d0f9a3..ec68eaea033 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3231,6 +3231,7 @@ enum {
 	CXT_FIXUP_INC_MIC_BOOST,
 	CXT_FIXUP_HEADPHONE_MIC_PIN,
 	CXT_FIXUP_HEADPHONE_MIC,
+	CXT_FIXUP_GPIO1,
 };
 
 static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
@@ -3375,6 +3376,15 @@ static const struct hda_fixup cxt_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cxt_fixup_headphone_mic,
 	},
+	[CXT_FIXUP_GPIO1] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			{ 0x01, AC_VERB_SET_GPIO_MASK, 0x01 },
+			{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01 },
+			{ 0x01, AC_VERB_SET_GPIO_DATA, 0x01 },
+			{ }
+		},
+	},
 };
 
 static const struct snd_pci_quirk cxt5051_fixups[] = {
@@ -3384,6 +3394,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = {
 
 static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1),
 	SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410),
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 7ea0245fc6b..50173d412ac 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -937,6 +937,14 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
 	}
 
 	/*
+	 * always configure channel mapping, it may have been changed by the
+	 * user in the meantime
+	 */
+	hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca,
+				   channels, per_pin->chmap,
+				   per_pin->chmap_set);
+
+	/*
 	 * sizeof(ai) is used instead of sizeof(*hdmi_ai) or
 	 * sizeof(*dp_ai) to avoid partial match/update problems when
 	 * the user switches between HDMI/DP monitors.
@@ -947,20 +955,10 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec,
 			    "pin=%d channels=%d\n",
 			    pin_nid,
 			    channels);
-		hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca,
-					   channels, per_pin->chmap,
-					   per_pin->chmap_set);
 		hdmi_stop_infoframe_trans(codec, pin_nid);
 		hdmi_fill_audio_infoframe(codec, pin_nid,
 					    ai.bytes, sizeof(ai));
 		hdmi_start_infoframe_trans(codec, pin_nid);
-	} else {
-		/* For non-pcm audio switch, setup new channel mapping
-		 * accordingly */
-		if (per_pin->non_pcm != non_pcm)
-			hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca,
-						   channels, per_pin->chmap,
-						   per_pin->chmap_set);
 	}
 
 	per_pin->non_pcm = non_pcm;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0e303b99a47..8ad554312b6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2819,6 +2819,15 @@ static void alc269_fixup_hweq(struct hda_codec *codec,
 	alc_write_coef_idx(codec, 0x1e, coef | 0x80);
 }
 
+static void alc269_fixup_headset_mic(struct hda_codec *codec,
+				       const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		spec->parse_flags |= HDA_PINCFG_HEADSET_MIC;
+}
+
 static void alc271_fixup_dmic(struct hda_codec *codec,
 			      const struct hda_fixup *fix, int action)
 {
@@ -3496,6 +3505,15 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec,
 	}
 }
 
+static void alc290_fixup_mono_speakers(struct hda_codec *codec,
+				       const struct hda_fixup *fix, int action)
+{
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		/* Remove DAC node 0x03, as it seems to be
+		   giving mono output */
+		snd_hda_override_wcaps(codec, 0x03, 0);
+}
+
 enum {
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -3507,6 +3525,7 @@ enum {
 	ALC271_FIXUP_DMIC,
 	ALC269_FIXUP_PCM_44K,
 	ALC269_FIXUP_STEREO_DMIC,
+	ALC269_FIXUP_HEADSET_MIC,
 	ALC269_FIXUP_QUANTA_MUTE,
 	ALC269_FIXUP_LIFEBOOK,
 	ALC269_FIXUP_AMIC,
@@ -3519,9 +3538,11 @@ enum {
 	ALC269_FIXUP_HP_GPIO_LED,
 	ALC269_FIXUP_INV_DMIC,
 	ALC269_FIXUP_LENOVO_DOCK,
+	ALC286_FIXUP_SONY_MIC_NO_PRESENCE,
 	ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT,
 	ALC269_FIXUP_DELL1_MIC_NO_PRESENCE,
 	ALC269_FIXUP_DELL2_MIC_NO_PRESENCE,
+	ALC269_FIXUP_DELL3_MIC_NO_PRESENCE,
 	ALC269_FIXUP_HEADSET_MODE,
 	ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC,
 	ALC269_FIXUP_ASUS_X101_FUNC,
@@ -3535,6 +3556,7 @@ enum {
 	ALC283_FIXUP_CHROME_BOOK,
 	ALC282_FIXUP_ASUS_TX300,
 	ALC283_FIXUP_INT_MIC,
+	ALC290_FIXUP_MONO_SPEAKERS,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -3603,6 +3625,10 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc269_fixup_stereo_dmic,
 	},
+	[ALC269_FIXUP_HEADSET_MIC] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc269_fixup_headset_mic,
+	},
 	[ALC269_FIXUP_QUANTA_MUTE] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc269_fixup_quanta_mute,
@@ -3712,6 +3738,15 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
 	},
+	[ALC269_FIXUP_DELL3_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
+	},
 	[ALC269_FIXUP_HEADSET_MODE] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_headset_mode,
@@ -3720,6 +3755,15 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_headset_mode_no_hp_mic,
 	},
+	[ALC286_FIXUP_SONY_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
+	},
 	[ALC269_FIXUP_ASUS_X101_FUNC] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc269_fixup_x101_headset_mic,
@@ -3804,6 +3848,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST
 	},
+	[ALC290_FIXUP_MONO_SPEAKERS] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc290_fixup_mono_speakers,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -3845,6 +3895,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS),
 	SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -3867,6 +3918,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101),
+	SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
 	SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
@@ -3952,6 +4004,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC269_FIXUP_STEREO_DMIC, .name = "alc269-dmic"},
 	{.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"},
 	{.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"},
+	{.id = ALC269_FIXUP_HEADSET_MIC, .name = "headset-mic"},
 	{.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"},
 	{.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"},
 	{.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"},
@@ -4569,6 +4622,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
+	SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_ASUS_MODE4),
+	SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_ASUS_MODE4),
 	SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT),
 	SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2),
 	SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 4f255dfee45..f59a321a6d6 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -4845,6 +4845,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 			if ((err = hdsp_get_iobox_version(hdsp)) < 0)
 				return err;
 		}
+		memset(&hdsp_version, 0, sizeof(hdsp_version));
 		hdsp_version.io_type = hdsp->io_type;
 		hdsp_version.firmware_rev = hdsp->firmware_rev;
 		if ((err = copy_to_user(argp, &hdsp_version, sizeof(hdsp_version))))
diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c
index c02405cc007..5810a0603f2 100644
--- a/sound/soc/blackfin/bf6xx-i2s.c
+++ b/sound/soc/blackfin/bf6xx-i2s.c
@@ -88,6 +88,7 @@ static int bfin_i2s_hw_params(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_FORMAT_S8:
 		param.spctl |= 0x70;
 		sport->wdsize = 1;
+		break;
 	case SNDRV_PCM_FORMAT_S16_LE:
 		param.spctl |= 0xf0;
 		sport->wdsize = 2;
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 8af04343cc1..259d1ac4492 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -349,6 +349,9 @@ static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol,
 	val = ucontrol->value.integer.value[0];
 	val2 = ucontrol->value.integer.value[1];
 
+	if (val >= ARRAY_SIZE(st_table) || val2 >= ARRAY_SIZE(st_table))
+		return -EINVAL;
+
 	err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m);
 	if (err < 0)
 		return err;
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index b8ba0adacfc..80555d7551e 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -1225,13 +1225,18 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
 	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
 	struct device *dev = codec->dev;
 	bool apply_fir, apply_iir;
-	int req, status;
+	unsigned int req;
+	int status;
 
 	dev_dbg(dev, "%s: Enter.\n", __func__);
 
 	mutex_lock(&drvdata->anc_lock);
 
 	req = ucontrol->value.integer.value[0];
+	if (req >= ARRAY_SIZE(enum_anc_state)) {
+		status = -EINVAL;
+		goto cleanup;
+	}
 	if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR &&
 		req != ANC_APPLY_IIR) {
 		dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n",
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 41cdd164297..8dbcacd44e6 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -1863,7 +1863,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol,
 	struct max98095_pdata *pdata = max98095->pdata;
 	int channel = max98095_get_eq_channel(kcontrol->id.name);
 	struct max98095_cdata *cdata;
-	int sel = ucontrol->value.integer.value[0];
+	unsigned int sel = ucontrol->value.integer.value[0];
 	struct max98095_eq_cfg *coef_set;
 	int fs, best, best_val, i;
 	int regmask, regsave;
@@ -2016,7 +2016,7 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol,
 	struct max98095_pdata *pdata = max98095->pdata;
 	int channel = max98095_get_bq_channel(codec, kcontrol->id.name);
 	struct max98095_cdata *cdata;
-	int sel = ucontrol->value.integer.value[0];
+	unsigned int sel = ucontrol->value.integer.value[0];
 	struct max98095_biquad_cfg *coef_set;
 	int fs, best, best_val, i;
 	int regmask, regsave;
diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
index 651ce092367..c91eba504f9 100644
--- a/sound/soc/codecs/pcm1681.c
+++ b/sound/soc/codecs/pcm1681.c
@@ -270,7 +270,7 @@ MODULE_DEVICE_TABLE(of, pcm1681_dt_ids);
 static const struct regmap_config pcm1681_regmap = {
 	.reg_bits		= 8,
 	.val_bits		= 8,
-	.max_register		= ARRAY_SIZE(pcm1681_reg_defaults) + 1,
+	.max_register		= 0x13,
 	.reg_defaults		= pcm1681_reg_defaults,
 	.num_reg_defaults	= ARRAY_SIZE(pcm1681_reg_defaults),
 	.writeable_reg		= pcm1681_writeable_reg,
diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c
index 2a8eccf64c7..7613181123f 100644
--- a/sound/soc/codecs/pcm1792a.c
+++ b/sound/soc/codecs/pcm1792a.c
@@ -188,7 +188,7 @@ MODULE_DEVICE_TABLE(of, pcm1792a_of_match);
 static const struct regmap_config pcm1792a_regmap = {
 	.reg_bits		= 8,
 	.val_bits		= 8,
-	.max_register		= 24,
+	.max_register		= 23,
 	.reg_defaults		= pcm1792a_reg_defaults,
 	.num_reg_defaults	= ARRAY_SIZE(pcm1792a_reg_defaults),
 	.writeable_reg		= pcm1792a_writeable_reg,
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 6e3f269243e..64ad84d8a30 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -674,6 +674,8 @@ static const struct snd_soc_dapm_route intercon[] = {
 	/* Left Input */
 	{"Left Line1L Mux", "single-ended", "LINE1L"},
 	{"Left Line1L Mux", "differential", "LINE1L"},
+	{"Left Line1R Mux", "single-ended", "LINE1R"},
+	{"Left Line1R Mux", "differential", "LINE1R"},
 
 	{"Left Line2L Mux", "single-ended", "LINE2L"},
 	{"Left Line2L Mux", "differential", "LINE2L"},
@@ -690,6 +692,8 @@ static const struct snd_soc_dapm_route intercon[] = {
 	/* Right Input */
 	{"Right Line1R Mux", "single-ended", "LINE1R"},
 	{"Right Line1R Mux", "differential", "LINE1R"},
+	{"Right Line1L Mux", "single-ended", "LINE1L"},
+	{"Right Line1L Mux", "differential", "LINE1L"},
 
 	{"Right Line2R Mux", "single-ended", "LINE2R"},
 	{"Right Line2R Mux", "differential", "LINE2R"},
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 8b50e5958de..01daf655e20 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -530,6 +530,7 @@ static int hp_supply_event(struct snd_soc_dapm_widget *w,
 				hubs->hp_startup_mode);
 			break;
 		}
+		break;
 
 	case SND_SOC_DAPM_PRE_PMD:
 		snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1,
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index c6b743978d5..6b81d0ce2c4 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -936,7 +936,7 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	ssi_private->ssi_phys = res.start;
 
 	ssi_private->irq = irq_of_parse_and_map(np, 0);
-	if (ssi_private->irq == NO_IRQ) {
+	if (ssi_private->irq == 0) {
 		dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
 		return -ENXIO;
 	}
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index a3d60d4bea4..a2fd7321b5a 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -112,7 +112,7 @@ static int imx_mc13783_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	if (machine_is_mx31_3ds()) {
+	if (machine_is_mx31_3ds() || machine_is_mx31moboard()) {
 		imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4,
 			IMX_AUDMUX_V2_PTCR_SYN,
 			IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) |
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index 46c5b4fdfc5..ca1be1d9dcf 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -62,7 +62,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
 	struct device_node *ssi_np, *codec_np;
 	struct platform_device *ssi_pdev;
 	struct i2c_client *codec_dev;
-	struct imx_sgtl5000_data *data;
+	struct imx_sgtl5000_data *data = NULL;
 	int int_port, ext_port;
 	int ret;
 
@@ -128,7 +128,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
 		goto fail;
 	}
 
-	data->codec_clk = devm_clk_get(&codec_dev->dev, NULL);
+	data->codec_clk = clk_get(&codec_dev->dev, NULL);
 	if (IS_ERR(data->codec_clk)) {
 		ret = PTR_ERR(data->codec_clk);
 		goto fail;
@@ -172,6 +172,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
 	return 0;
 
 fail:
+	if (data && !IS_ERR(data->codec_clk))
+		clk_put(data->codec_clk);
 	if (ssi_np)
 		of_node_put(ssi_np);
 	if (codec_np)
@@ -185,6 +187,7 @@ static int imx_sgtl5000_remove(struct platform_device *pdev)
 	struct imx_sgtl5000_data *data = platform_get_drvdata(pdev);
 
 	snd_soc_unregister_card(&data->card);
+	clk_put(data->codec_clk);
 
 	return 0;
 }
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index f58bcd85c07..57d6941676f 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -600,19 +600,17 @@ static int imx_ssi_probe(struct platform_device *pdev)
 	ssi->fiq_params.dma_params_rx = &ssi->dma_params_rx;
 	ssi->fiq_params.dma_params_tx = &ssi->dma_params_tx;
 
-	ret = imx_pcm_fiq_init(pdev, &ssi->fiq_params);
-	if (ret)
-		goto failed_pcm_fiq;
+	ssi->fiq_init = imx_pcm_fiq_init(pdev, &ssi->fiq_params);
+	ssi->dma_init = imx_pcm_dma_init(pdev);
 
-	ret = imx_pcm_dma_init(pdev);
-	if (ret)
-		goto failed_pcm_dma;
+	if (ssi->fiq_init && ssi->dma_init) {
+		ret = ssi->fiq_init;
+		goto failed_pcm;
+	}
 
 	return 0;
 
-failed_pcm_dma:
-	imx_pcm_fiq_exit(pdev);
-failed_pcm_fiq:
+failed_pcm:
 	snd_soc_unregister_component(&pdev->dev);
 failed_register:
 	release_mem_region(res->start, resource_size(res));
@@ -628,8 +626,11 @@ static int imx_ssi_remove(struct platform_device *pdev)
 	struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	struct imx_ssi *ssi = platform_get_drvdata(pdev);
 
-	imx_pcm_dma_exit(pdev);
-	imx_pcm_fiq_exit(pdev);
+	if (!ssi->dma_init)
+		imx_pcm_dma_exit(pdev);
+
+	if (!ssi->fiq_init)
+		imx_pcm_fiq_exit(pdev);
 
 	snd_soc_unregister_component(&pdev->dev);
 
diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h
index fb1616ba8c5..560c40fc9eb 100644
--- a/sound/soc/fsl/imx-ssi.h
+++ b/sound/soc/fsl/imx-ssi.h
@@ -211,6 +211,8 @@ struct imx_ssi {
 	struct imx_dma_data filter_data_rx;
 	struct imx_pcm_fiq_params fiq_params;
 
+	int fiq_init;
+	int dma_init;
 	int enabled;
 };
 
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index daa78a0095f..4a07f717969 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -1,6 +1,6 @@
 config SND_OMAP_SOC
 	tristate "SoC Audio for the Texas Instruments OMAP chips"
-	depends on (ARCH_OMAP && DMA_OMAP) || (ARCH_ARM && COMPILE_TEST)
+	depends on (ARCH_OMAP && DMA_OMAP) || (ARM && COMPILE_TEST)
 	select SND_DMAENGINE_PCM
 
 config SND_OMAP_SOC_DMIC
@@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810
 
 config SND_OMAP_SOC_RX51
 	tristate "SoC Audio support for Nokia RX-51"
-	depends on SND_OMAP_SOC && ARCH_ARM && (MACH_NOKIA_RX51 || COMPILE_TEST)
+	depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST)
 	select SND_OMAP_SOC_MCBSP
 	select SND_SOC_TLV320AIC3X
 	select SND_SOC_TPA6130A2
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index 9cc6986a8cf..5dd87f4c919 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -220,8 +220,8 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv,
 void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv,
 			       struct rsnd_mod *mod,
 			       enum rsnd_reg reg);
-#define rsnd_is_gen1(s)		((s)->info->flags & RSND_GEN1)
-#define rsnd_is_gen2(s)		((s)->info->flags & RSND_GEN2)
+#define rsnd_is_gen1(s)		(((s)->info->flags & RSND_GEN_MASK) == RSND_GEN1)
+#define rsnd_is_gen2(s)		(((s)->info->flags & RSND_GEN_MASK) == RSND_GEN2)
 
 /*
  *	R-Car ADG
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 4d0561312f3..1a38be0d0ca 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1380,7 +1380,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order)
 				return -ENODEV;
 
 			list_add(&cpu_dai->dapm.list, &card->dapm_list);
-			snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai);
 		}
 
 		if (cpu_dai->driver->probe) {
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index c17c14c394d..b2949aed1ac 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1949,7 +1949,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 				w->active ? "active" : "inactive");
 
 	list_for_each_entry(p, &w->sources, list_sink) {
-		if (p->connected && !p->connected(w, p->sink))
+		if (p->connected && !p->connected(w, p->source))
 			continue;
 
 		if (p->connect)
@@ -3495,6 +3495,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 		if (!w) {
 			dev_err(dapm->dev, "ASoC: Failed to create %s widget\n",
 				dai->driver->playback.stream_name);
+			return -ENOMEM;
 		}
 
 		w->priv = dai;
@@ -3513,6 +3514,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 		if (!w) {
 			dev_err(dapm->dev, "ASoC: Failed to create %s widget\n",
 				dai->driver->capture.stream_name);
+			return -ENOMEM;
 		}
 
 		w->priv = dai;
diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c
index d0323a693ba..999550bbad4 100644
--- a/sound/usb/usx2y/us122l.c
+++ b/sound/usb/usx2y/us122l.c
@@ -262,7 +262,9 @@ static int usb_stream_hwdep_mmap(struct snd_hwdep *hw,
 	}
 
 	area->vm_ops = &usb_stream_hwdep_vm_ops;
-	area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP;
+	area->vm_flags |= VM_DONTDUMP;
+	if (!read)
+		area->vm_flags |= VM_DONTEXPAND;
 	area->vm_private_data = us122l;
 	atomic_inc(&us122l->mmap_count);
 out:
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index 63fb5219f0f..6234a51625b 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -299,19 +299,6 @@ static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y,
 	usX2Y_clients_stop(usX2Y);
 }
 
-static void usX2Y_error_sequence(struct usX2Ydev *usX2Y,
-				 struct snd_usX2Y_substream *subs, struct urb *urb)
-{
-	snd_printk(KERN_ERR
-"Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n"
-"Most probably some urb of usb-frame %i is still missing.\n"
-"Cause could be too long delays in usb-hcd interrupt handling.\n",
-		   usb_get_current_frame_number(usX2Y->dev),
-		   subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out",
-		   usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame);
-	usX2Y_clients_stop(usX2Y);
-}
-
 static void i_usX2Y_urb_complete(struct urb *urb)
 {
 	struct snd_usX2Y_substream *subs = urb->context;
@@ -328,12 +315,9 @@ static void i_usX2Y_urb_complete(struct urb *urb)
 		usX2Y_error_urb_status(usX2Y, subs, urb);
 		return;
 	}
-	if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF)))
-		subs->completed_urb = urb;
-	else {
-		usX2Y_error_sequence(usX2Y, subs, urb);
-		return;
-	}
+
+	subs->completed_urb = urb;
+
 	{
 		struct snd_usX2Y_substream *capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE],
 			*playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK];
diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c
index f2a1acdc4d8..814d0e887c6 100644
--- a/sound/usb/usx2y/usx2yhwdeppcm.c
+++ b/sound/usb/usx2y/usx2yhwdeppcm.c
@@ -244,13 +244,8 @@ static void i_usX2Y_usbpcm_urb_complete(struct urb *urb)
 		usX2Y_error_urb_status(usX2Y, subs, urb);
 		return;
 	}
-	if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF)))
-		subs->completed_urb = urb;
-	else {
-		usX2Y_error_sequence(usX2Y, subs, urb);
-		return;
-	}
 
+	subs->completed_urb = urb;
 	capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE];
 	capsubs2 = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE + 2];
 	playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK];